b86d4e4a8d
This CL contains three changes as a preparation for adding audio send streams to the send-side BWE: 1. Audio packets are passed through the pacer with high priority. This is needed to be able to set transport sequence numbers on the packets. 2. A feedback observer is passed to the audio stream's rtcp receiver so that the BWE can get notified of any BWE feedback being received on the audio feedback channel. 3. Support for the transport sequence number header extension is added to audio send streams. BUG=webrtc:5263,webrtc:5307 R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1479023002 . Cr-Commit-Position: refs/heads/master@{#10909}
100 lines
3.2 KiB
C++
100 lines
3.2 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_SEND_STREAM_H_
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#define WEBRTC_AUDIO_SEND_STREAM_H_
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#include <string>
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#include <vector>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/config.h"
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#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
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#include "webrtc/stream.h"
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#include "webrtc/transport.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// WORK IN PROGRESS
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// This class is under development and is not yet intended for for use outside
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// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
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// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
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class AudioSendStream : public SendStream {
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public:
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struct Stats {
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// TODO(solenberg): Harmonize naming and defaults with receive stream stats.
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uint32_t local_ssrc = 0;
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int64_t bytes_sent = 0;
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int32_t packets_sent = 0;
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int32_t packets_lost = -1;
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float fraction_lost = -1.0f;
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std::string codec_name;
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int32_t ext_seqnum = -1;
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int32_t jitter_ms = -1;
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int64_t rtt_ms = -1;
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int32_t audio_level = -1;
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float aec_quality_min = -1.0f;
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int32_t echo_delay_median_ms = -1;
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int32_t echo_delay_std_ms = -1;
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int32_t echo_return_loss = -100;
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int32_t echo_return_loss_enhancement = -100;
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bool typing_noise_detected = false;
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};
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struct Config {
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Config() = delete;
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explicit Config(Transport* send_transport)
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: send_transport(send_transport) {}
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std::string ToString() const;
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// Receive-stream specific RTP settings.
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struct Rtp {
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std::string ToString() const;
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// Sender SSRC.
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uint32_t ssrc = 0;
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// RTP header extensions used for the sent stream.
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std::vector<RtpExtension> extensions;
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// RTCP CNAME, see RFC 3550.
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std::string c_name;
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} rtp;
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// Transport for outgoing packets. The transport is expected to exist for
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// the entire life of the AudioSendStream and is owned by the API client.
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Transport* send_transport = nullptr;
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// Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
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// components.
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// TODO(solenberg): Remove when VoiceEngine channels are created outside
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// of Call.
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int voe_channel_id = -1;
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// Ownership of the encoder object is transferred to Call when the config is
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// passed to Call::CreateAudioSendStream().
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// TODO(solenberg): Implement, once we configure codecs through the new API.
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// rtc::scoped_ptr<AudioEncoder> encoder;
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int cng_payload_type = -1; // pt, or -1 to disable Comfort Noise Generator.
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int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
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};
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// TODO(solenberg): Make payload_type a config property instead.
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virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
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uint32_t duration_ms) = 0;
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virtual Stats GetStats() const = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_SEND_STREAM_H_
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